FXS dial plan is: { L: 911+ | L: <4=>x+ | x+ | +x+ | *x+ | xxx+ }. This setup and dial plan uses PSTN for numbers that start with 911 or 4 (and drops the 4), and V2 for all other numbers. The V1_DID account registered on the FXO tab does not work for outbound or inbound calls.

My problem might be solved if I knew how to use the VoIP account registered on the FXO tab. Then I could register V1_DID on the FXS tab and V2 on the FXO tab.

[Side note: I was expecting the HT813 to have 2 profile tabs in the browser setup screen, like HT812, then use a dial plan a bit like a billion modem: { <4=>x+=@pstn | <2=>x+=@Profile2 | x+ | etc. } where initial 4 uses PSTN and initial 2 uses the second VOIP profile.)

If one of the 2 profiles can only be used to dial in, then this should be clear.

HT813 is one of the few current products with FXO, and it is a real pity it does not offer the software functionality of multiple dial-out VoIP accounts that is so common. Why have the unusual ability to dial in SIP on FXO (how is this is useful?), without providing a more common dial out functionality.

FXO is used to connect a PSTN (POTS) line and convert to SIP. This allows the use of a traditional phone line with a SIP system for those who need to have this type of connectivity. Calls can be made in and out.

The FXS aspect is typically used to connect analog phones to a SIP server. It may also be used to connect and incoming SIP Trunk to a legacy analog PBX.

The very first sentence on the product page indicate this - “The HT813 is an analog telephone adapter that features 1 analog telephone FXS port and 1 PSTN line FXO port.”

Quintum and Patton make a two port FXO gateway, and there may be others, but GS does not offer such a product as the next step up is a 4 port. However, the cost of the GS 4-port may be quite favorable to the other two brands which may make it a consideration.

I have used routers with an fxo port and an FXS port with multiple dial out sip accounts for about 20 years. This is the first product I have owned that has only 1 dial out sip profile on the FXS port.

I think you are saying that the second sip profile can only be used to dial out (not dial in, as dial in on the FXO VoIP account does not work), that this dial out can only be used by someone dialing in on the pots/pstn line as a type of call forwarding. I do not need this.

I still think Grandstream’s advertising is misleading. Advertising should state clearly that there is only 1 sip dial-out profile available for the phone connected to the fxs port, and that the profiles are tied to the respective port. The profiles or ports are not “integrated”, contrary to the manual.

I also think that you have misunderstood what I need. I do -not- need multiple fxo ports. I only need 1 fxo and 1 fxs, like what I have used for 20 years (my last device was Billion 7404). I do need at least 2 dial out sip profiles on the 1 fxs port, with the account selected by the dial plan. All the routers I mentioned in my previous message have this functionality.

Unfortunately I have already purchased this ata and I’m feeling disapointed, do not want to spend more money.

Grandstand has a unique product. It’s a real pity the software is limited to 1 dial out profile on the FXS port.

I am not indicating anything about dial-out or dial-in. I am defining what the functionalities of FXO and FXS are. Dialing out has no real meaning without knowing how each port will be utilized and connected to what device… Each has a profile by which to connect to a SIP server.

I took a look at the router you are used to and assuming I found the correct manual, the 7404 and 813 are two different animals. The 7404 is a DSL router that also has provision for a common phone line (no DSL filter). The dial code usage allows you to dial out via VoIP/SIP or the phone line from the 2 phone ports. The profiles/accounts are related to the 2 FXS (phone) ports the device supports. There is no FXO port per se as there is no profile/account association to same. It appears that the dial code merely connects the phone port to the line port.

Take a look at the HT503 and then resources and the hop-on, hop-off document. Perhaps this defines the desire?

The billion 7404 VoIP models (eg VGxx, VNxx) are indeed a different device. They are a modem router with built in ata. The Grandstand is just an ATA. Whether the Billion is setup for the VoIP to exit to the SIP server using the provided ADSL, USB, WAN/EWAN or LAN ports (via a separate modem) is largely irrelevant to the ATA VoIP functionality, as this related to the modem router functionality

But you are wrong about the VoIP functionality. The Billion 7404 VoIP models have an FXO and 2 FXS ports and support multiple VoIP accounts (profiles) - I’ve used up to 3 on each FXS Port. Similarly the equivalent Fritz Box devices support up to 20 SIP accounts on 2 FXS ports. The equivalent Vigor devices support up to 6 VoIP accounts on 2 FXS ports.

This is the functionality:

A VoIP account is tied to each FXS Port. This is the only dial in number on that port, and is the default dial out account.

A phone connected to 1 FXS Port can dial out on additional VOIP accounts using the dial plan formula, with encoding similar to Grandstream’s ‘L’ (up to 20 accounts on Fritz Box, 6 on Vigor and about the same on Billion 7404 Vxxx)

A phone connected to an FXS port can dial out on the PSTN line in various ways including a dial plan formula similar to Grandstream ‘L’

Calls received on the PSTN line can be received on either or both FXS ports.

A VoIP account is tied to each FXS Port. This is the only dial in number on that port, and is the default dial out account.

A phone connected to 1 FXS Port can dial out on additional VOIP accounts using the dial plan formula, with encoding similar to Grandstream’s ‘L’ (up to 20 accounts on Fritz Box, 6 on Vigor and about the same on Billion 7404 Vxxx)

A phone connected to an FXS port can dial out on the PSTN line in various ways including a dial plan formula similar to Grandstream ‘L’

Calls received on the PSTN line can be received on either or both FXS ports.:

Exactly as I understand it. So as I stated, there is no profile for the FXO. The FXO you are used to is not SIP (connecting the PSTN to a SIP device). It is a phone port (PSTN). The ATAs will not really mimic what the devices you are used to using.

As there is only one FXS. It will however take the PSTN call to the available FXS.

This device is typically used to connect to a SIP server (IP-PBX) so that both the FXO and FXS ports are registered to same. In this manner any phone connected to the PBX can have a call routed in or out the FXO (PSTN), which may be beneficial if desired to have emergency calls routed,out to a local responder rather than a national response center and possibly routed in to a given extension for a return call. The FXS acts as an extension to the PBX and can call any other extension on the PBX. In the event of a power outage, the FXO (PSTN) and FXS are connected automatically (lifeline mode) so that the analog phone connected to the FXS can still make calls out and take calls in.

When you have an integrated device such as yours or from Flying Voice, Draytek or other, it is typically FXS only so that you can connect analog phones to a provider and then for various reason - least cost routing, emergency, fax, etc - can direct out via the attached PSTN port. It can be a pretty effective poor man’s PBX and especially so if the provider supports some of the other telephony features -call waiting, forwarding, etc…

I am not aware of any standalone gateway or ATA that will allow profile switching via dial -code. If anyone might, IMO, perhaps Patton or Cisco